Remove unused stuff from AudioFrame:

webrtc/modules/audio_coding/neteq/sync_buffer.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 61 matching lines @@
 
 void SyncBuffer::GetNextAudioInterleaved(size_t requested_len,
                                          AudioFrame* output) {
   RTC_DCHECK(output);
   const size_t samples_to_read = std::min(FutureLength(), requested_len);
   output->Reset();
   const size_t tot_samples_read =
       ReadInterleavedFromIndex(next_index_, samples_to_read, output->data_);
   const size_t samples_read_per_channel = tot_samples_read / Channels();
   next_index_ += samples_read_per_channel;
-  output->interleaved_ = true;
   output->num_channels_ = Channels();
   output->samples_per_channel_ = samples_read_per_channel;
 }
 
 void SyncBuffer::IncreaseEndTimestamp(uint32_t increment) {
   end_timestamp_ += increment;
 }
 
 void SyncBuffer::Flush() {
   Zeros(Size());
   next_index_ = Size();
   end_timestamp_ = 0;
   dtmf_index_ = 0;
 }
 
 void SyncBuffer::set_next_index(size_t value) {
   // Cannot set |next_index_| larger than the size of the buffer.
   next_index_ = std::min(value, Size());
 }
 
 void SyncBuffer::set_dtmf_index(size_t value) {
   // Cannot set |dtmf_index_| larger than the size of the buffer.
   dtmf_index_ = std::min(value, Size());
 }
 
 }  // namespace webrtc